1. Field of the Invention
The present invention relates to a signal processing apparatus for a microphone array comprised of a plurality of microphones arranged in a given space, a signal processing method for the microphone array, and a microphone array system.
2. Description of the Related Art
Conventionally, array processing has been proposed in which delays are added to signals of sound received by a microphone array comprised of a plurality of microphones arranged in a given space, and then the signals are summed so that directivity is given to the microphone array (Japanese Laid-Open Patent Publication (Kokai) No. H09-140000, and “Acoustic System and Digital Processing” co-authored by Toshiro Oga, Yoshio Yamazaki, and Yutaka Kaneda, The Institute of Electronics, Information and Communication Engineers (issued on Mar. 25, 1995), see Pages 181 to 186). Such array processing is referred to as “delay-and-sum processing” or “DS (Delay-and-Sum) processing.”
The principle of the DS processing will be summarized below.
In general, a microphone array system is comprised of a microphone array of M (M is a positive integer not less than 2) microphones MICi (i is a positive integer from 1 to M), delay devices that give delays Di to audio signals xsi(t) output from the respective microphones, and an adder that sums the delayed sound signals xsi(t−Di). For simplicity, it is assumed that the microphone array working as sound receivers is implemented by an equally-spaced linear microphone array comprised of M microphones arranged at regular intervals in a line.
By giving suitable delays Di to sound signals xsi(t) output from the respective microphones, it is possible to correct for the time lags between sounds reaching the respective microphones from the intended direction θL (the direction in which the microphone array is desired to have directivity) so that the sounds can be in phase. On the other hand, sounds reaching the respective microphones from directions other than the intended direction θL cannot be in phase by the above delay processing. Thus, when the delayed sound signals xsi(t−Di) are summed, the signals being in phase are emphasized, but the signals not being in phase are not so emphasized. As a result, the microphone array has such a directional characteristic as to be highly sensitive to sound coming from the intended direction θL.
According to the above-mentioned “Acoustic System and Digital Processing”, the directional characteristic of the microphone array system obtained by the above described DS processing can be expressed as below. First, the amplitude ratio of the array processing output y(t) and the array input xi(t), i.e. the array gain G can be expressed by the following equations (1) and (2):G=|sin(ΩM/2)/sin(ΩM/2)|  (1)where Ω=2πfd(sin θL−sin θ)/c  (2)
f: Frequency of the sound signal
d: Distance between microphones
θL: Intended direction
θ: Direction from which sound comes
c: Sound velocity
The directional characteristic of the microphone array system before the array gain G becomes zero (or a sufficiently low gain) is referred to as a mainlobe; the array gain G becomes zero for the first time on the condition that the following equation (3) using the above equation (1) is satisfied:ΩM/2=π  (3)
When θL=0, the angle θ1 (mainlobe width) at which the array gain G becomes zero for the first time is expressed by the following equation (4) using the above equations (2) and (3):θ1=sin−1(c/fdM)  (4)
As is evident from the above equation (4), the mainlobe width decreases as the frequency f, the distance between microphones d, and the number of microphones M increase.
According to the above-mentioned “Acoustic System and Digital Processing”, the microphone array system has the following properties regarding the directional characteristic, which apply to array types other than linear arrays:
(1) When large values are selected as the number of microphones M and the distance between microphones d, and the array length Md is set to be long, a sharp directional characteristic in the intended direction can be realized.
(2) The mainlobe width depends on the frequency (i.e., the higher the frequency, the sharper the directional characteristic).
(3) When the distance between microphones d is less than c/2f, no spatial loopback of the mainlobe occurs.
It should be noted that the applicant has found no prior art related to the present invention except for Laid-Open Patent Publication (Kokai) Nos. H09-140000, H06-202627, and H09-251044 (corresponding to U.S. Pat. No. 5,960,373) as well as the above-mentioned “Acoustic System and Digital Processing”.
The array length of the microphone array as a whole must be long so as to obtain a sharp directional characteristic for a low frequency band due to the above described properties of the DS microphone array system, and this has been a hindrance to the downsizing of the microphone array. Also, when a compact microphone array is used, a satisfactorily sharp directional characteristic cannot be realized, and hence there is the problem that sound signals in a low frequency band are buried in other sound signals (noise) coming from the surroundings.